Debugging SIP problems

Hi all,

I’m trying to set up an Asterisk server, but am struggling slightly - largely because I’ve never worked with Asterisk or Linux before!

I’ve managed to get the server installed, and I’ve got two accounts that work fine when used with soft IP phones (eg SJPhone).

However, I’ve also got 3 Cisco 7911 IP phones that aren’t so happy. I did get these phones SIP enabled and connected, but when I dial a working softphone extension number, I get a “Reorder” error message and a repeating dial tone.

I don’t mind spending time researching this, but to be honest I don’t know where to start! Is there some sort of log that I can take a look at, and if so how? The FreePBX server doesn’t seem to give me access to any logs, and the server itself is showing a command line.

Many thanks in advance guys,


at a command prompt type asterisk -vvvr (v’s) not w’S)

that will give you an instant display of whats going on.

can the cisco’s call from 1 to the other?

can you dial any test’s echo speaking clock etc etc

and to get out do ctrl c.

im still a bit of a noob, and am still learning (been using voip/asterisk for 6 years now!)

Thanks Tabpole, that looks like a good start - but what do I do about the fact that all the info disappears off the top of the screen? I’m at a pure command prompt, not inside any kind of GUI box.